handle_request_register not a local domain error

I am trying to have my Linksys SPA962 connect to my asterisk server. now since the phone did connect to a simple trixbox installation in a rather simple way, i assume my problem lies in the configuration of my Asterisk server.

this is the error in my Asterisk CLI:
[Oct 15 11:13:24] NOTICE[8346]: chan_sip.c:17790 handle_request_register: Registration from ‘“LDAPTST03” sip:LDAPTST03@’ failed for ‘’ - Not a local domain

when using X-Lite i can enter a Domain name as which is should connect, and by doing so X-Lite will register and be able to make calls. now the SPA692 does not have such an option, thus i thought i’d make an entry for the IP range 10.10.10.x into the sip.conf, though this does not work either.

Anyone that can tell me what i should change?

My Sip.conf

; SIP Configuration example for Asterisk
; SIP dial strings
; In the dialplan (extensions.conf) you can use several
; syntaxes for dialing SIP devices.
; SIP/devicename
; SIP/username@domain (SIP uri)
; SIP/username[:password[:md5secret[]]]@host[:port]
; SIP/devicename/extension
; Devicename
; devicename is defined as a peer in a section below.
; username@domain
; Call any SIP user on the Internet
; (Don’t forget to enable DNS SRV records if you want to use this)
; devicename/extension
; If you define a SIP proxy as a peer below, you may call
; SIP/proxyhostname/user or SIP/user@proxyhostname
; where the proxyhostname is defined in a section below
; This syntax also works with ATA’s with FXO ports
; SIP/username[:password[:md5secret[]]]@host[:port]
; This form allows you to specify password or md5secret and authname
; without altering any authentication data in config.
; Examples:
; SIP/*98@mysipproxy
; SIP/sales:topsecret::account02@domain.com:5062
; SIP/12345678::bc53f0ba8ceb1ded2b70e05c3f91de4f:myname@
; All of these dial strings specify the SIP request URI.
; In addition, you can specify a specific To: header by adding an
; exclamation mark after the dial string, like
; SIP/sales@mysipproxy!sales@edvina.net
; CLI Commands
; -------------------------------------------------------------
; Useful CLI commands to check peers/users:
; sip show peers Show all SIP peers (including friends)
; sip show users Show all SIP users (including friends)
; sip show registry Show status of hosts we register with
; sip set debug Show all SIP messages
; sip reload Reload configuration file
; Active SIP peers will not be reconfigured

; ** Deprecated configuration options **
; The “call-limit” configuation option is deprecated. It still works in
; this version of Asterisk, but will disappear in the next version.
; You are encouraged to use the dialplan groupcount functionality
; to enforce call limits instead of using this channel-specific method.
; You can still set limits per device in sip.conf or in a database by using
; “setvar” to set variables that can be used in the dialplan for various limits.

context=default ; Default context for incoming calls
;allowguest=no ; Allow or reject guest calls (default is yes)
match_auth_username=yes ; if available, match user entry using the
; ‘username’ field from the authentication line
; instead of the From: field.

allowoverlap=no ; Disable overlap dialing support. (Default is yes)
;allowtransfer=no ; Disable all transfers (unless enabled in peers or users)
; Default is enabled
realm=walbeekgroep.local ; Realm for digest authentication
; defaults to “asterisk”. If you set a system name in
; asterisk.conf, it defaults to that system name
; Realms MUST be globally unique according to RFC 3261
; Set this to your host name or domain name
bindport=5060 ; UDP Port to bind to (SIP standard port is 5060)
; bindport is the local UDP port that Asterisk will listen on
bindaddr= ; IP address to bind to ( binds to all)

; Note that the TCP and TLS support for chan_sip is currently considered
; experimental. Since it is new, all of the related configuration options are
; subject to change in any release. If they are changed, the changes will
; be reflected in this sample configuration file, as well as in the UPGRADE.txt file.
tcpenable=yes ; Enable server for incoming TCP connections (default is yes)
tcpbindaddr= ; IP address for TCP server to bind to ( binds to all interfaces)
; Optionally add a port number, (default is port 5060)

;tlsenable=no ; Enable server for incoming TLS (secure) connections (default is no)
;tlsbindaddr= ; IP address for TLS server to bind to ( binds to all interfaces)
; Optionally add a port number, (default is port 5061)
; Remember that the IP address must match the common name (hostname) in the
; certificate, so you don’t want to bind a TLS socket to multiple IP addresses.

;tlscertfile=asterisk.pem ; Certificate file (*.pem only) to use for TLS connections
; default is to look for “asterisk.pem” in current directory

; If the server your connecting to uses a self signed certificate
; you should have their certificate installed here so the code can
; verify the authenticity of their certificate.

; A directory full of CA certificates. The files must be named with
; the CA subject name hash value.
; (see man SSL_CTX_load_verify_locations for more info)

; If set to yes, don’t verify the servers certificate when acting as
; a client. If you don’t have the server’s CA certificate you can
; set this and it will connect without requiring tlscafile to be set.
; Default is no.

; A string specifying which SSL ciphers to use or not use
; A list of valid SSL cipher strings can be found at:
; http://www.openssl.org/docs/apps/ciphers.html#CIPHER_STRINGS

srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; Note: Asterisk only uses the first host
; in SRV records
; Disabling DNS SRV lookups disables the
; ability to place SIP calls based on domain
; names to some other SIP users on the Internet

domain=walbeekgroep.local ; Set default domain for this host
; If configured, Asterisk will only allow
; INVITE and REFER to non-local domains
; Use “sip show domains” to list local domains
;pedantic=yes ; Enable checking of tags in headers,
; international character conversions in URIs
; and multiline formatted headers for strict
; SIP compatibility (defaults to “no”)

; See qos.tex or Quality of Service section of asterisk.pdf for a description of these parameters.
;tos_sip=cs3 ; Sets TOS for SIP packets.
;tos_audio=ef ; Sets TOS for RTP audio packets.
;tos_video=af41 ; Sets TOS for RTP video packets.
;tos_text=af41 ; Sets TOS for RTP text packets.

;cos_sip=3 ; Sets 802.1p priority for SIP packets.
;cos_audio=5 ; Sets 802.1p priority for RTP audio packets.
;cos_video=4 ; Sets 802.1p priority for RTP video packets.
;cos_text=3 ; Sets 802.1p priority for RTP text packets.

;maxexpiry=3600 ; Maximum allowed time of incoming registrations
; and subscriptions (seconds)
;minexpiry=60 ; Minimum length of registrations/subscriptions (default 60)
;defaultexpiry=120 ; Default length of incoming/outgoing registration
;qualifyfreq=60 ; Qualification: How often to check for the
; host to be up in seconds
; Set to low value if you use low timeout for
; NAT of UDP sessions
;notifymimetype=text/plain ; Allow overriding of mime type in MWI NOTIFY
;buggymwi=no ; Cisco SIP firmware doesn’t support the MWI RFC
; fully. Enable this option to not get error messages
; when sending MWI to phones with this bug.
;vmexten=voicemail ; dialplan extension to reach mailbox sets the
; Message-Account in the MWI notify message
; defaults to “asterisk”
;disallow=all ; First disallow all codecs
allow=all ; Allow codecs in order of preference
;allow=ilbc ; see doc/rtp-packetization for framing options
; This option specifies a preference for which music on hold class this channel
; should listen to when put on hold if the music class has not been set on the
; channel with Set(CHANNEL(musicclass)=whatever) in the dialplan, and the peer
; channel putting this one on hold did not suggest a music class.
; This option may be specified globally, or on a per-user or per-peer basis.
; This option specifies which music on hold class to suggest to the peer channel
; when this channel places the peer on hold. It may be specified globally or on
; a per-user or per-peer basis.
;language=en ; Default language setting for all users/peers
; This may also be set for individual users/peers
;relaxdtmf=yes ; Relax dtmf handling
;trustrpid = no ; If Remote-Party-ID should be trusted
;sendrpid = yes ; If Remote-Party-ID should be sent
;progressinband=never ; If we should generate in-band ringing always
; use ‘never’ to never use in-band signalling, even in cases
; where some buggy devices might not render it
; Valid values: yes, no, never Default: never
;useragent=Asterisk PBX ; Allows you to change the user agent string
; The default user agent string also contains the Asterisk
; version. If you don’t want to expose this, change the
; useragent string.
;sdpsession=Asterisk PBX ; Allows you to change the SDP session name string, (s=)
; Like the useragent parameter, the default user agent string
; also contains the Asterisk version.
;sdpowner=root ; Allows you to change the username field in the SDP owner string, (o=)
; This field MUST NOT contain spaces
;promiscredir = no ; If yes, allows 302 or REDIR to non-local SIP address
; Note that promiscredir when redirects are made to the
; local system will cause loops since Asterisk is incapable
; of performing a “hairpin” call.
;usereqphone = no ; If yes, “;user=phone” is added to uri that contains
; a valid phone number
;dtmfmode = rfc2833 ; Set default dtmfmode for sending DTMF. Default: rfc2833
; Other options:
; info : SIP INFO messages (application/dtmf-relay)
; shortinfo : SIP INFO messages (application/dtmf)
; inband : Inband audio (requires 64 kbit codec -alaw, ulaw)
; auto : Use rfc2833 if offered, inband otherwise

;compactheaders = yes ; send compact sip headers.
videosupport=yes ; Turn on support for SIP video. You need to turn this on
; in the this section to get any video support at all.
; You can turn it off on a per peer basis if the general
; video support is enabled, but you can’t enable it for
; one peer only without enabling in the general section.
;maxcallbitrate=384 ; Maximum bitrate for video calls (default 384 kb/s)
; Videosupport and maxcallbitrate is settable
; for peers and users as well
;callevents=no ; generate manager events when sip ua
; performs events (e.g. hold)
alwaysauthreject = no ; When an incoming INVITE or REGISTER is to be rejected,
; for any reason, always reject with ‘401 Unauthorized’
; instead of letting the requester know whether there was
; a matching user or peer for their request

;g726nonstandard = yes ; If the peer negotiates G726-32 audio, use AAL2 packing
; order instead of RFC3551 packing order (this is required
; for Sipura and Grandstream ATAs, among others). This is
; contrary to the RFC3551 specification, the peer should
; be negotiating AAL2-G726-32 instead :frowning:
;outboundproxy=proxy.provider.domain ; send outbound signaling to this proxy, not directly to the devices
;outboundproxy=proxy.provider.domain:8080 ; send outbound signaling to this proxy, not directly to the devices
;outboundproxy=proxy.provider.domain,force ; Send ALL outbound signalling to proxy, ignoring route: headers
;matchexterniplocally = yes ; Only substitute the externip or externhost setting if it matches
; your localnet setting. Unless you have some sort of strange network
; setup you will not need to enable this.

; If regcontext is specified, Asterisk will dynamically create and destroy a
; NoOp priority 1 extension for a given peer who registers or unregisters with
; us and have a “regexten=” configuration item.
; Multiple contexts may be specified by separating them with ‘&’. The
; actual extension is the ‘regexten’ parameter of the registering peer or its
; name if ‘regexten’ is not provided. If more than one context is provided,
; the context must be specified within regexten by appending the desired
; context after ‘@’. More than one regexten may be supplied if they are
; separated by ‘&’. Patterns may be used in regexten.
regextenonqualify=yes ; Default “no”
; If you have qualify on and the peer becomes unreachable
; this setting will enforce inactivation of the regexten
; extension for the peer
;--------------------------- SIP timers ----------------------------------------------------
; These timers are used primarily in INVITE transactions.
; The default for Timer T1 is 500 ms or the measured run-trip time between
; Asterisk and the device if you have qualify=yes for the device.
;t1min=100 ; Minimum roundtrip time for messages to monitored hosts
; Defaults to 100 ms
;timert1=500 ; Default T1 timer
; Defaults to 500 ms or the measured round-trip
; time to a peer (qualify=yes).
;timerb=32000 ; Call setup timer. If a provisional response is not received
; in this amount of time, the call will autocongest
; Defaults to 64*timert1

;--------------------------- RTP timers ----------------------------------------------------
; These timers are currently used for both audio and video streams. The RTP timeouts
; are only applied to the audio channel.
; The settings are settable in the global section as well as per device
;rtptimeout=60 ; Terminate call if 60 seconds of no RTP or RTCP activity
; on the audio channel
; when we’re not on hold. This is to be able to hangup
; a call in the case of a phone disappearing from the net,
; like a powerloss or grandma tripping over a cable.
;rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP or RTCP activity
; on the audio channel
; when we’re on hold (must be > rtptimeout)
;rtpkeepalive= ; Send keepalives in the RTP stream to keep NAT open
; (default is off - zero)

;--------------------------- SIP Session-Timers (RFC 4028)------------------------------------
; SIP Session-Timers provide an end-to-end keep-alive mechanism for active SIP sessions.
; This mechanism can detect and reclaim SIP channels that do not terminate through normal
; signaling procedures. Session-Timers can be configured globally or at a user/peer level.
; The operation of Session-Timers is driven by the following configuration parameters:
; * session-timers - Session-Timers feature operates in the following three modes:
; originate : Request and run session-timers always
; accept : Run session-timers only when requested by other UA
; refuse : Do not run session timers in any case
; The default mode of operation is ‘accept’.
; * session-expires - Maximum session refresh interval in seconds. Defaults to 1800 secs.
; * session-minse - Minimum session refresh interval in seconds. Defualts to 90 secs.
; * session-refresher - The session refresher (uac|uas). Defaults to ‘uas’.

;--------------------------- SIP DEBUGGING ---------------------------------------------------
;sipdebug = yes ; Turn on SIP debugging by default, from
; the moment the channel loads this configuration
recordhistory=yes ; Record SIP history by default
; (see sip history / sip no history)
;dumphistory=yes ; Dump SIP history at end of SIP dialogue
; SIP history is output to the DEBUG logging channel

;--------------------------- STATUS NOTIFICATIONS (SUBSCRIPTIONS) ----------------------------
; You can subscribe to the status of extensions with a “hint” priority
; (See extensions.conf.sample for examples)
; chan_sip support two major formats for notifications: dialog-info and SIMPLE
; You will get more detailed reports (busy etc) if you have a call counter enabled
; for a device.
; If you set the busylevel, we will indicate busy when we have a number of calls that
; matches the busylevel treshold.
; For queues, you will need this level of detail in status reporting, regardless
; if you use SIP subscriptions. Queues and manager use the same internal interface
; for reading status information.
; Note: Subscriptions does not work if you have a realtime dialplan and use the
; realtime switch.
;allowsubscribe=no ; Disable support for subscriptions. (Default is yes)
;subscribecontext = default ; Set a specific context for SUBSCRIBE requests
; Useful to limit subscriptions to local extensions
; Settable per peer/user also
;notifyringing = yes ; Notify subscriptions on RINGING state (default: no)
;notifyhold = yes ; Notify subscriptions on HOLD state (default: no)
; Turning on notifyringing and notifyhold will add a lot
; more database transactions if you are using realtime.
;callcounter = yes ; Enable call counters on devices. This can be set per
; device too.
;counteronpeer = yes ; Apply call counting on peers only. This will improve
; status notification when you are using type=friend
; Inbound calls, that really apply to the user part
; of a friend will now be added to and compared with
; the peer counter instead of applying two call counters,
; one for the peer and one for the user.
; “sip show inuse” will only show active calls on
; the peer side of a “type=friend” object if this
; setting is turned on.

;----------------------------------------- T.38 FAX PASSTHROUGH SUPPORT -----------------------
; This setting is available in the [general] section as well as in device configurations.
; Setting this to yes, enables T.38 fax (UDPTL) passthrough on SIP to SIP calls, provided
; both parties have T38 support enabled in their Asterisk configuration
; This has to be enabled in the general section for all devices to work. You can then
; disable it on a per device basis.
; T.38 faxing only works in SIP to SIP calls, with no local or agent channel being used.
; t38pt_udptl = yes ; Default false
;----------------------------------------- OUTBOUND SIP REGISTRATIONS ------------------------
; Asterisk can register as a SIP user agent to a SIP proxy (provider)
; Format for the register statement is:
; register => [transport://]user[:secret[]]@host[:port][/extension]
; If no extension is given, the ‘s’ extension is used. The extension needs to
; be defined in extensions.conf to be able to accept calls from this SIP proxy
; (provider).
; host is either a host name defined in DNS or the name of a section defined
; below.
; A similar effect can be achieved by adding a “callbackextension” option in a peer section.
; this is equivalent to having the following line in the general section:
; register => username:secret@host/callbackextension
; and more readable because you don’t have to write the parameters in two places
; (note that the “port” is ignored - this is a bug that should be fixed).
; Examples:
;register => 1234:password@mysipprovider.com
; This will pass incoming calls to the ‘s’ extension
;register => 2345:password@sip_proxy/1234
; Register 2345 at sip provider ‘sip_proxy’. Calls from this provider
; connect to local extension 1234 in extensions.conf, default context,
; unless you configure a [sip_proxy] section below, and configure a
; context.
; Tip 1: Avoid assigning hostname to a sip.conf section like [provider.com]
; Tip 2: Use separate type=peer and type=user sections for SIP providers
; (instead of type=friend) if you have calls in both directions

;registertimeout=20 ; retry registration calls every 20 seconds (default)
;registerattempts=10 ; Number of registration attempts before we give up
; 0 = continue forever, hammering the other server
; until it accepts the registration
; Default is 0 tries, continue forever

;----------------------------------------- NAT SUPPORT ------------------------
; WARNING: SIP operation behind a NAT is tricky and you really need
; to read and understand well the following section.
; When Asterisk is behind a NAT device, the “local” address (and port) that
; a socket is bound to has different values when seen from the inside or
; from the outside of the NATted network. Unfortunately this address must
; be communicated to the outside (e.g. in SIP and SDP messages), and in
; order to determine the correct value Asterisk needs to know:
; + whether it is talking to someone “inside” or “outside” of the NATted network.
; This is configured by assigning the “localnet” parameter with a list
; of network addresses that are considered “inside” of the NATted network.
; Multiple entries are allowed, e.g. a reasonable set is the following:
; localnet= ; RFC 1918 addresses
; localnet= ; Also RFC1918
; localnet= ; Another RFC1918 with CIDR notation
; localnet= ; Zero conf local network
; + the “externally visible” address and port number to be used when talking
; to a host outside the NAT. This information is derived by one of the
; following (mutually exclusive) config file parameters:
; a. “externip = hostname[:port]” specifies a static address[:port] to
; be used in SIP and SDP messages.
; The hostname is looked up only once, when [re]loading sip.conf .
; If a port number is not present, use the “bindport” value (which is
; not guaranteed to work correctly, because a NAT box might remap the
; port number as well as the address).
; This approach can be useful if you have a NAT device where you can
; configure the mapping statically. Examples:
; externip = ; use this address.
; externip = ; use this address and port.
; externip = mynat.my.org:12600 ; Public address of my nat box.
; b. “externhost = hostname[:port]” is similar to “externip” except
; that the hostname is looked up every “externrefresh” seconds
; (default 10s). This can be useful when your NAT device lets you choose
; the port mapping, but the IP address is dynamic.
; Beware, you might suffer from service disruption when the name server
; resolution fails. Examples:
; externhost=foo.dyndns.net ; refreshed periodically
; externrefresh=180 ; change the refresh interval
; c. “stunaddr = stun.server[:port]” queries the STUN server specified
; as an argument to obtain the external address/port.
; Queries are also sent periodically every “externrefresh” seconds
; (as a side effect, sending the query also acts as a keepalive for
; the state entry on the nat box):
; stunaddr = foo.stun.com:3478
; externrefresh = 15
; Note that at the moment all these mechanism work only for the SIP socket.
; The IP address discovered with externip/externhost/STUN is reused for
; media sessions as well, but the port numbers are not remapped so you
; may still experience problems.
; NOTE 1: in some cases, NAT boxes will use different port numbers in
; the internal<->external mapping. In these cases, the “externip” and
; “externhost” might not help you configure addresses properly, and you
; really need to use STUN.
; NOTE 2: when using “externip” or “externhost”, the address part is
; also used as the external address for media sessions.
; If you use “stunaddr”, STUN queries will be sent to the same server
; also from media sockets, and this should permit a correct mapping of
; the port numbers as well.
; In addition to the above, Asterisk has an additional “nat” parameter to
; address NAT-related issues in incoming SIP or media sessions.
; In particular, depending on the 'nat= ’ settings described below, Asterisk
; may override the address/port information specified in the SIP/SDP messages,
; and use the information (sender address) supplied by the network stack instead.
; However, this is only useful if the external traffic can reach us.
; The following settings are allowed (both globally and in individual sections):
; nat = no ; default. Use NAT mode only according to RFC3581 (;rport)
; nat = yes ; Always ignore info and assume NAT
; nat = never ; Never attempt NAT mode or RFC3581 support
; nat = route ; route = Assume NAT, don’t send rport
; ; (work around more UNIDEN bugs)

;----------------------------------- MEDIA HANDLING --------------------------------
; By default, Asterisk tries to re-invite the audio to an optimal path. If there’s
; no reason for Asterisk to stay in the media path, the media will be redirected.
; This does not really work with in the case where Asterisk is outside and have
; clients on the inside of a NAT. In that case, you want to set canreinvite=nonat
;canreinvite=yes ; Asterisk by default tries to redirect the
; RTP media stream (audio) to go directly from
; the caller to the callee. Some devices do not
; support this (especially if one of them is behind a NAT).
; The default setting is YES. If you have all clients
; behind a NAT, or for some other reason wants Asterisk to
; stay in the audio path, you may want to turn this off.

			; This setting also affect direct RTP
			; at call setup (a new feature in 1.4 - setting up the
			; call directly between the endpoints instead of sending
			; a re-INVITE).

;directrtpsetup=yes ; Enable the new experimental direct RTP setup. This sets up
; the call directly with media peer-2-peer without re-invites.
; Will not work for video and cases where the callee sends
; RTP payloads and fmtp headers in the 200 OK that does not match the
; callers INVITE. This will also fail if canreinvite is enabled when
; the device is actually behind NAT.

;canreinvite=nonat ; An additional option is to allow media path redirection
; (reinvite) but only when the peer where the media is being
; sent is known to not be behind a NAT (as the RTP core can
; determine it based on the apparent IP address the media
; arrives from).

;canreinvite=update ; Yet a third option… use UPDATE for media path redirection,
; instead of INVITE. This can be combined with ‘nonat’, as
; ‘canreinvite=update,nonat’. It implies ‘yes’.

;----------------------------------------- REALTIME SUPPORT ------------------------
; For additional information on ARA, the Asterisk Realtime Architecture,
; please read realtime.txt and extconfig.txt in the /doc directory of the
; source code.
rtcachefriends=yes ; Cache realtime friends by adding them to the internal list
; just like friends added from the config file only on a
; as-needed basis? (yes|no)

;rtsavesysname=yes ; Save systemname in realtime database at registration
; Default= no

rtupdate=yes ; Send registry updates to database using realtime? (yes|no)
; If set to yes, when a SIP UA registers successfully, the ip address,
; the origination port, the registration period, and the username of
; the UA will be set to database via realtime.
; If not present, defaults to ‘yes’.
;rtautoclear=yes ; Auto-Expire friends created on the fly on the same schedule
; as if it had just registered? (yes|no|)
; If set to yes, when the registration expires, the friend will
; vanish from the configuration until requested again. If set
; to an integer, friends expire within this number of seconds
; instead of the registration interval.

;ignoreregexpire=yes ; Enabling this setting has two functions:
; For non-realtime peers, when their registration expires, the
; information will not be removed from memory or the Asterisk database
; if you attempt to place a call to the peer, the existing information
; will be used in spite of it having expired
; For realtime peers, when the peer is retrieved from realtime storage,
; the registration information will be used regardless of whether
; it has expired or not; if it expires while the realtime peer
; is still in memory (due to caching or other reasons), the
; information will not be removed from realtime storage

;----------------------------------------- SIP DOMAIN SUPPORT ------------------------
; Incoming INVITE and REFER messages can be matched against a list of ‘allowed’
; domains, each of which can direct the call to a specific context if desired.
; By default, all domains are accepted and sent to the default context or the
; context associated with the user/peer placing the call.
; Domains can be specified using:
; domain=[,]
; Examples:
; domain=myasterisk.dom
; domain=customer.com,customer-context
; In addition, all the ‘default’ domains associated with a server should be
; added if incoming request filtering is desired.
; autodomain=yes
; To disallow requests for domains not serviced by this server:
; allowexternaldomains=no

; Add domain and configure incoming context
; for external calls to this domain
; Add IP address as local domain
; You can have several “domain” settings
;allowexternaldomains=no ; Disable INVITE and REFER to non-local domains
; Default is yes
autodomain=yes ; Turn this on to have Asterisk add local host
; name and local IP to domain list.

fromdomain=walbeekgroep.local ; When making outbound SIP INVITEs to
; non-peers, use your primary domain “identity”
; for From: headers instead of just your IP
; address. This is to be polite and
; it may be a mandatory requirement for some
; destinations which do not have a prior
; account relationship with your server.

;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a
; SIP channel. Defaults to “no”. An enabled jitterbuffer will
; be used only if the sending side can create and the receiving
; side can not accept jitter. The SIP channel can accept jitter,
; thus a jitterbuffer on the receive SIP side will be used only
; if it is forced and enabled.

; jbforce = no ; Forces the use of a jitterbuffer on the receive side of a SIP
; channel. Defaults to “no”.

; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds.

; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
; resynchronized. Useful to improve the quality of the voice, with
; big jumps in/broken timestamps, usually sent from exotic devices
; and programs. Defaults to 1000.

; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a SIP
; channel. Two implementations are currently available - “fixed”
; (with size always equals to jbmaxsize) and “adaptive” (with
; variable size, actually the new jb of IAX2). Defaults to fixed.

; jblog = no ; Enables jitterbuffer frame logging. Defaults to “no”.

; Global credentials for outbound calls, i.e. when a proxy challenges your
; Asterisk server for authentication. These credentials override
; any credentials in peer/register definition if realm is matched.
; This way, Asterisk can authenticate for outbound calls to other
; realms. We match realm on the proxy challenge and pick an set of

; credentials from this list
; Syntax:
; auth = :@
; auth = #@
; Example:
; You may also add auth= statements to [peer] definitions
; Peer auth= override all other authentication settings if we match on realm

; Users and peers have different settings available. Friends have all settings,
; since a friend is both a peer and a user
; User config options: Peer configuration:
; -------------------- -------------------
; context context
; callingpres callingpres
; permit permit
; deny deny
; secret secret
; md5secret md5secret
; dtmfmode dtmfmode
; canreinvite canreinvite
; nat nat
; callgroup callgroup
; pickupgroup pickupgroup
; language language
; allow allow
; disallow disallow
; insecure insecure
; trustrpid trustrpid
; progressinband progressinband
; promiscredir promiscredir
; useclientcode useclientcode
; accountcode accountcode
; setvar setvar
; callerid callerid
; amaflags amaflags
; call-limit call-limit (deprecated)
; callcounter callcounter
; allowoverlap allowoverlap
; allowsubscribe allowsubscribe
; allowtransfer allowtransfer
; subscribecontext subscribecontext
; videosupport videosupport
; maxcallbitrate maxcallbitrate
; rfc2833compensate mailbox
; session-timers busylevel
; session-expires
; session-minse template
; session-refresher fromdomain
; regexten
; fromuser
; host
; port
; qualify
; defaultip
; defaultuser
; rtptimeout
; rtpholdtimeout
; sendrpid
; outboundproxy
; rfc2833compensate
; callbackextension
; registertrying
; session-timers
; session-expires
; session-minse
; session-refresher
; timert1
; timerb
; qualifyfreq

; For incoming calls only. Example: FWD (Free World Dialup)
; We match on IP address of the proxy for incoming calls
; since we can not match on username (caller id)

;type=peer ; we only want to call out, not be called
;defaultuser=yourusername ; Authentication user for outbound proxies
;fromuser=yourusername ; Many SIP providers require this!
;usereqphone=yes ; This provider requires “;user=phone” on URI
;callcounter=yes ; Enable call counter
;busylevel=2 ; Signal busy at 2 or more calls
;outboundproxy=proxy.provider.domain ; send outbound signaling to this proxy, not directly to the peer
;port=80 ; The port number we want to connect to on the remote side
; Also used as “defaultport” in combination with “defaultip” settings

;— sample definition for a provider
;fromuser=4015552299 ; how your provider knows you
;callbackextension=123 ; Register with this server and require calls coming back to this extension

; Definitions of locally connected SIP devices
; type = user a device that authenticates to us by “from” field to place calls
; type = peer a device we place calls to or that calls us and we match by host
; type = friend two configurations (peer+user) in one
; For device names, we recommend using only a-z, numerics (0-9) and underscore
; For local phones, type=friend works most of the time
; If you have one-way audio, you probably have NAT problems.
; If Asterisk is on a public IP, and the phone is inside of a NAT device
; you will need to configure nat option for those phones.
; Also, turn on qualify=yes to keep the nat session open
; Because you might have a large number of similar sections, it is generally
; convenient to use templates for the common parameters, and add them
; the the various sections. Examples are below, and we can even leave
; the templates uncommented as they will not harm:

basic-options ; a template

natted-phone ; another template inheriting basic-options

public-phone ; another template inheriting basic-options

my-codecs ; a template for my preferred codecs

ulaw-phone ; and another one for ulaw-only

; and finally instantiate a few phones
; 2133
; secret = peekaboo
; 2134
; secret = not_very_secret
; 2136
; secret = not_very_secret_either
; …

; Standard configurations not using templates look like this:
;context=from-sip ; Where to start in the dialplan when this phone calls
;callerid=John Doe <1234> ; Full caller ID, to override the phones config
; on incoming calls to Asterisk
;host= ; we have a static but private IP address
; No registration allowed
;nat=no ; there is not NAT between phone and Asterisk
;canreinvite=yes ; allow RTP voice traffic to bypass Asterisk
;dtmfmode=info ; either RFC2833 or INFO for the BudgeTone
;call-limit=1 ; permit only 1 outgoing call and 1 incoming call at a time
; from the phone to asterisk (deprecated)
; 1 for the explicit peer, 1 for the explicit user,
; remember that a friend equals 1 peer and 1 user in
; memory
; There is no combined call counter for a “friend”
; so there’s currently no way in sip.conf to limit
; to one inbound or outbound call per phone. Use
; the group counters in the dial plan for that.
;mailbox=1234@default ; mailbox 1234 in voicemail context “default”
;disallow=all ; need to disallow=all before we can use allow=
;allow=ulaw ; Note: In user sections the order of codecs
; listed with allow= does NOT matter!
;allow=g723.1 ; Asterisk only supports g723.1 pass-thru!
;allow=g729 ; Pass-thru only unless g729 license obtained
;callingpres=allowed_passed_screen ; Set caller ID presentation
; See README.callingpres for more information

; Turn off silence suppression in X-Lite (“Transmit Silence”=YES)!
; Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed
;regexten=1234 ; When they register, create extension 1234
;callerid=“Jane Smith” <5678>
;host=dynamic ; This device needs to register
;nat=yes ; X-Lite is behind a NAT router
;canreinvite=no ; Typically set to NO if behind NAT
;allow=gsm ; GSM consumes far less bandwidth than ulaw
;mailbox=1234@default,1233@default ; Subscribe to status of multiple mailboxes
;registertrying=yes ; Send a 100 Trying when the device registers.

;type=friend ; Friends place calls and receive calls
;context=from-sip ; Context for incoming calls from this user
;subscribecontext=localextensions ; Only allow SUBSCRIBE for local extensions
;language=de ; Use German prompts for this user
;host=dynamic ; This peer register with us
;dtmfmode=inband ; Choices are inband, rfc2833, or info
;defaultip= ; IP used until peer registers
;mailbox=1234@context,2345 ; Mailbox(-es) for message waiting indicator
;subscribemwi=yes ; Only send notifications if this phone
; subscribes for mailbox notification
;vmexten=voicemail ; dialplan extension to reach mailbox
; sets the Message-Account in the MWI notify message
; defaults to global vmexten which defaults to “asterisk”
;allow=ulaw ; dtmfmode=inband only works with ulaw or alaw!

;type=friend ; Friends place calls and receive calls
;context=from-sip ; Context for incoming calls from this user
;host=dynamic ; This peer register with us
;dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info
;defaultuser=polly ; Username to use in INVITE until peer registers
; Normally you do NOT need to set this parameter
;allow=ulaw ; dtmfmode=inband only works with ulaw or alaw!
;progressinband=no ; Polycom phones don’t work properly with “never”

;insecure=port ; Allow matching of peer by IP address without
; matching port number
;insecure=invite ; Do not require authentication of incoming INVITEs
;insecure=port,invite ; (both)
;qualify=1000 ; Consider it down if it’s 1 second to reply
; Helps with NAT session
; qualify=yes uses default value
;qualifyfreq=60 ; Qualification: How often to check for the
; host to be up in seconds
; Set to low value if you use low timeout for
; NAT of UDP sessions
; Call group and Pickup group should be in the range from 0 to 63
;callgroup=1,3-4 ; We are in caller groups 1,3,4
;pickupgroup=1,3-5 ; We can do call pick-p for call group 1,3,4,5
;defaultip= ; IP address to use if peer has not registered
;deny= ; ACL: Control access to this account based on IP address

;qualify=200 ; Qualify peer is no more than 200ms away
;nat=yes ; This phone may be natted
; Send SIP and RTP to the IP address that packet is
; received from instead of trusting SIP headers
;host=dynamic ; This device registers with us
;canreinvite=no ; Asterisk by default tries to redirect the
; RTP media stream (audio) to go directly from
; the caller to the callee. Some devices do not
; support this (especially if one of them is
; behind a NAT).
;defaultip= ; IP address to use until registration
;defaultuser=goran ; Username to use when calling this device before registration
; Normally you do NOT need to set this parameter
;setvar=CUSTID=5678 ; Channel variable to be set for all calls from this device

;rfc2833compensate=yes ; Compensate for pre-1.4 DTMF transmission from another Asterisk machine.
; You must have this turned on or DTMF reception will work improperly.

callerid=“STDUSR01” <0001>

callerid=“STDUSR02” <0002>

forgive the double post XD most likely due to the large post my firefox crashed and since i didnt see the message i reposted it.

btw, might be best to make it continue in this thread, since the thread title is better for the googlers with the same error :smile:

shamelss bump :blush: